Audio signal rate adaptation method and system

ABSTRACT

A system for adapting a transmission rate of an audio signal transmitted over a TDM communications link from a source to a destination includes a first variable-depth storage component and a first processor at the source, and a second variable-length storage component and a second processor at the destination. The first storage component stores the audio signal, and the first processor selectively withdraws the signal from the first storage component. The first processor encapsulates the signal into packets and transmits the packets via the TDM link. The second processor receives the packets, removes the audio signal, and deposits the signal into the second variable-depth storage component. The first processor selectively withdraws the audio signal as a function of the amount of data in the first storage component, and the second processor selectively withdraws the audio signal as a function of an amount of data in the second storage component.

CROSS-REFERENCE TO RELATED APPLICATIONS

[0001] This application is related to the following U.S. applications,of common assignee, from which priority is claimed, and the contents ofwhich are incorporated herein in their entirety by reference:

[0002] “Audio Signal Rate Adaptation Method And System,” U.S.Provisional Patent Application Serial No. 60/370,286, filed Apr. 5,2002.

STATEMENT REGARDING FEDERALLY SPONSORED RESEARCH

[0003] Not Applicable

REFERENCE TO MICROFICHE APPENDIX

[0004] Not Applicable

BACKGROUND

[0005] The professional quality (i.e., program) audio that is producedin a studio for a commercial radio broadcast must often be relayed to aremote transmitter site for over-the-air transmission. One techniqueknown in the art for relaying such program audio over significantdistances is to sample the audio to produce a digital data stream, thentransmit the digital data on a TDM link such as a T1 or E1 digitalcircuit in the Public Switched Telephone Network (PSTN), Microwave linksand other media. In order to preserve the integrity of the programaudio, the sampled data is often transmitted in a linear range, i.e.,not compressed in any way.

[0006] A common format for the digital audio data is the AudioEngineering Society/European Broadcast Union (AES/EBU) digital audiostandard. AES/EBU is a bit-serial communications protocol fortransmitting digital audio data through a single transmission line. Thisstandard allows for two channels of audio data, up to 24 bits persample, channel status bits for communication control and statusinformation, and some error detection capabilities. Clocking information(i.e., sample rate), embedded in the AES/EBU bit stream, is recovered atthe receiving end of the transmission path. The AES/EBU standardspecifies the use of 32 kHz, 44.1 kHz, or 48 kHz sample rates.

[0007]FIG. 1 shows a prior art system for transmitting program audio inAES/EBU format from a studio site 12 to a remote transmitter site 14. Areceive buffer 16 receives the AES/EBU data from the production source,the recovered clock and data are then provided to an asynchronous samplerate converter (ASRC) 18. The ASRC 18 converts the timing of the inputAES/EBU data to conform to the TDM backplane clock associated with thestudio multiplexor. The ASRC provides the data to the CPU 20 timed tothe backplane clock. The PLL 24 derives the ASRC output clock, as wellas the clock to the CPU 20, from the TDM backplane clock. The CPU 20processes the AES/EBU audio data and passes this audio data to a commonmodule (CM) 22, which implements (among other things) the T1 TDMfunctionality at the studio site 12. The CM 22 encapsulates the AES/EBUaudio data into the T1 TDM link destined for the remote transmitter site14.

[0008] A second CM 26 at the remote transmitter site 14 receives the T1data from the studio site 12, extracts the timing from the T1 signal andthe AES/EBU audio data from the T1 time slots, and passes the AES/EBUdata to the CPU 28. The PLL 32 provides a clock to the CPU 28 that issynchronized to the TDM backplane clock. The CPU 28, after suitableprocessing, provides the AES/EBU data to a transmit buffer 30, whichconditions and drives the AES/EBU audio to the appropriate destinationwithin the transmitter site 14.

[0009] In the system of FIG. 1, all components after the ASRC 18 in thetransmission path are synchronized to the TDM clock, so the audio datatransfer from the studio site 12 to the transmitter site 14 occurs at afairly constant rate. The ASRC 18, however, while ensuring a relativelysmooth data transfer, can deteriorate the quality of the audio beingtransferred. For example, the ASRC FIR filter is a sharp-cutoff low-passfilter, which can produce “ringing” in waveforms that contain fasttransitions. This can cause the peak FM deviation to be exceeded.Further, the finite arithmetic representation of the samples within theFIR algorithm requires one or more requantization steps, which introduceadditional noise into the system. The additional hardware necessary toimplement the ASRC 18 adds to the overall cost of the system. Theprogram audio is typically relayed to the transmitter in a linear range,i.e., without compression, to avoid similar deterioration in the audioquality. It is therefore undesirable to have the ASRC 18 in the datapath.

SUMMARY OF THE INVENTION

[0010] The foregoing and other objects are achieved by the inventionwhich in one aspect comprises a system for adapting a transmission rateof an audio signal transmitted over a time division multiplexedcommunications link from a source to a destination. The system includesa first variable-depth storage component for receiving and temporarilystoring the audio signal. The system also includes a first processor forselectively withdrawing the audio signal from the first variable-depthstorage component, for encapsulating the audio signal into packetscompatible with the time division multiplexed communication link, andfor transmitting the packets over the time division multiplexedcommunications link. Also included in the system is a second processorfor receiving the packets from the time division multiplexedcommunications link, for removing the audio signal from theencapsulating packets, and for depositing the audio signal into a secondvariable-depth storage component. The first processor selectivelywithdraws the audio signal as a predetermined function of an amount ofdata in the first variable-depth storage component, and the secondprocessor selectively withdraws the audio signal as a predeterminedfunction of an amount of data in the second variable-depth storagecomponent.

[0011] In another embodiment, the first variable-depth storage componentincludes a FIFO. In yet another embodiment, the second variable-depthstorage component includes a FIFO.

[0012] In another embodiment, the audio signal includes an AES/EBUdigital audio signal.

[0013] In another embodiment the first processor internally implementsthe first variable-depth storage component, and the second processorinternally implements the second variable-depth storage component.

[0014] In another embodiment, the second processor is clocked by aclocking signal derived from a signal source having a frequency that isvariable according to a control signal from the second processor. Thesecond processor monitors the second variable-length storage componentand adjusts the control signal so as to maintain the secondvariable-depth storage component approximately half full. In oneembodiment, the signal source includes a PLL. Yet another embodimentfurther includes a signal conditioner for receiving a signal from thesignal source and producing a clocking signal therefrom. In anotherembodiment, the signal conditioner is implemented with a FPGA device.

[0015] In another embodiment, the first processor being clocked by aclocking signal derived from a signal source having a frequency which isvariable according to a control signal derived from the time divisionmultiplexed communications link.

[0016] Another aspect of the invention comprises a method of adapting atransmission rate of an audio signal transmitted over a time divisionmultiplexed communications link from a source to a destination. Themethod includes encapsulating the audio signal within the time divisionmultiplexed communications link. The method further includes unpackingthe audio signal from the encapsulation within the time divisionmultiplexed communications link, and varying the transmission rate ofthe audio signal at the destination, after unpacking from the timedivision multiplexed communications link.

[0017] Another aspect of the invention comprises a method of adapting atransmission rate of an audio signal transmitted over a time divisionmultiplexed communications link from a source to a destination,including receiving and temporarily storing the audio signal in a firstvariable-depth storage component. The method further includesselectively withdrawing the audio signal from the first variable-depthstorage component as a predetermined function of an amount of data inthe first variable-depth storage component, and encapsulating the audiosignal into packets compatible with the time division multiplexedcommunication link. The encapsulated audio signal is then transmittedvia the packets over the time division multiplexed communications link.The method also includes receiving the packets from the time divisionmultiplexed communications link, and removing the audio signal from theencapsulating packets. The extracted audio signal is deposited into asecond variable-depth storage component, and the audio signal isselectively withdrawn as a predetermined function of an amount of datain the second variable-depth storage component.

[0018] Another embodiment further includes implementing the firstvariable-depth storage component with a FIFO. Another embodiment furtherincludes implementing the second variable-depth storage component with aFIFO.

[0019] One embodiment further includes adapting the audio signalincludes as an AES/EBU digital audio signal.

[0020] Yet another embodiment further includes implementing the firstvariable-depth storage component and the second variable-depth storagecomponent in code, running on one or more processors.

[0021] Another aspect of the invention comprises a system for adapting atransmission rate of an audio signal transmitted over a time divisionmultiplexed communications link from a source to a destination. Thesystem includes an encapsulator, including a first variable-depthstorage component for receiving and temporarily storing the audiosignal. The encapsulator (i) selectively withdraws the audio signal fromthe first variable-depth storage component, (ii) encapsulates the audiosignal into packets compatible with the time division multiplexedcommunication link, and (iii) transmits the packets over the timedivision multiplexed communications link. The system also includes anextractor for receiving the packets from the time division multiplexedcommunications link, for removing the audio signal from theencapsulating packets, and for depositing the audio signal into a secondvariable-depth storage component. The encapsulator selectively withdrawsthe audio signal as a predetermined function of an amount of data in thefirst variable-depth storage component, and the extractor selectivelywithdraws the audio signal as a predetermined function of an amount ofdata in the second variable-depth storage component.

BRIEF DESCRIPTION OF DRAWINGS

[0022] The foregoing and other objects of this invention, the variousfeatures thereof, as well as the invention itself, may be more fullyunderstood from the following description, when read together with theaccompanying drawings in which:

[0023]FIG. 1 shows a prior art system for transmitting program audio inAES/EBU format from a studio site to a remote transmitter site; and,

[0024]FIG. 2 shows one embodiment of a system for adapting the audiosignal rate in a TDM communications system.

DESCRIPTION OF THE PREFERRED EMBODIMENTS

[0025]FIG. 2 shows one embodiment of a system 100 for adapting the audiosignal rate in a TDM communications system. The system 100 relaysprogram audio in AES/EBU format from a studio site 102 to a remotetransmitter site 104. In general, the system 100 encapsulates theasynchronous AES/EBU audio data into a T1 TDM transmission withoutsample rate conversion, but performs any necessary rate adaptation atthe studio site 102 before being transmitted and at the remotetransmitter site 104, after the audio data has been extracted from theTDM transmission. The system 100 employs an “encapsulator” (i.e., asystem or apparatus for encapsulating audio into a TDM transmission) ator near the studio site (i.e., the source), and an “extractor” (i.e., asystem or apparatus for extracting the audio from a TDM transmission) ator near the remote transmitter site (i.e., the destination). FIG. 2shows one embodiment of the encapsulator and extractor, although otherembodiment may also be used.

[0026] A receive buffer 106 receives the AES/EBU data from theproduction source, conditions then provides the audio data to a CPU 108.The CPU 108 receives a clock from a PLL 112. The PLL 112 derives theclock to the CPU 108 and the PLL 112 is controlled from the internalFIFO depth in the CPU 108. The FIFO depth represents the amount ofincoming AES/EBU data from the AES/EBU receiver 106. In one embodimentof the system 100, the CPU 108 uses the clock recovered from the inputAES/EBU data stream to receive the AES/EBU data. In other embodiments,the CPU 108 handles the AES/EBU data with a derivative of the PLL clock112. The CPU 108 processes the AES/EBU audio data and passes this audiodata to a common module (CM) 110, which implements (among other things)the T1 TDM functionality at the studio site 102. The CM 110 encapsulatesthe AES/EBU audio data into the T1 TDM link destined for the remotetransmitter site 104. One of the processing functions of the CPU 108 isto packetize the data into a superframe per the various transmissionformats that are supported by the system 100. A frame packer in the CPU108 formats the data from the audio FIFO 109 into the 8-kHz framesrequired by the T1/E1. The CPU 108 also determines how many time slots,per unit time, should be filled in the T1 TDM transmission. Since theincoming AES/EBU audio data is asynchronous with respect to the T1transmission, the number of AES/EBU audio data filled per unit time inthe T1 transmission cannot remain constant. The CPU 108 thereforecontrols the amount of audio data and auxiliary data that is sent in therequired number of timeslots used in each T1 transmission.

[0027] A second CM 114 at the remote transmitter site 104 receives theT1 data from the studio site 102, extracts the AES/EBU audio data fromthe T1 time slots, and passes the AES/EBU data to the CPU 116 via theTDM bus timing. The CM 114 module from the T1 input extracts this TDMtiming. A framer algorithm in the CPU 116 synchronizes itself to thesuperframe structure of the data packet by looking for the incrementing4-bit pattern that exists in the first byte of each superframe toestablish the frame boundaries. The frame unpacker then separates theauxiliary data packets from the audio samples, and the audio samples aresent to the audio FIFO 118 in the CPU 116.

[0028] In one embodiment, the CPU 116 works in conjunction with a FIFO118 to process the AES/EBU data from the CM 114. The AES/EBU audio datafrom the CM 114 is clocked into the FIFO 118 with the TDM bus clock. TheAES/EBU data is clocked out of the CPU 114 with the clock 124 derived bythe PLL 120. The CPU 116 monitors the FIFO 118 and dynamically variesthe control signal 122 to the PLL 120 so as to maintain the FIFO 118 atapproximately half full. In operation, when the CPU 116 detects that theFIFO 118 is greater than half full, the CPU adjusts the control signal122 so as to increase the frequency of the first clock 124, thusincreasing the rate at which the FIFO 118 is emptied. When the CPU 116detects that the FIFO 118 is less than half full, the CPU 116 adjuststhe control signal 122 so as to decrease the frequency of the firstclock 124, thus decreasing the rate at which the FIFO 118 is emptied. Ina way, the PLL 120 that produces the first clock 124 is locked to thehalf full flag of the FIFO 118, which is indirectly related to theoriginal AES/EBU timing at the input of the studio 102. Thus, the PLL120 is essentially locked to the original AES/EBU timing at the studio102.

[0029] The CPU 116 provides the AES/EBU data that has been removed fromthe FIFO 118 to an AES/EBU transmitter 119, which conditions and drivesthe AES/EBU audio to the appropriate destination within the transmittersite 104.

[0030] A related system is described in U.S. Pat. No. 5,818,769,entitled Dynamically Variable Digital Delay Line, and is incorporated byreference herein in its entirety. The '769 patent describes an elasticdigital delay line that implements the elasticity by varying the outputrate of the delay line. A PLL controls the output rate of the delay lineas a function of a control signal tied to the desired amount of delay,similar to the present invention that controls the output rate of theFIFO 118 as a function of the amount of data in the FIFO 118. Many ofthe same techniques taught in the '769 patent may also be used toimplement various embodiments of the present invention. For example, the'769 patent teaches first adjusting the PLL according to a courseresolution, and then refining the adjustment according to a fineresolution. The system 100 may also initially adjust the PLL 120according to a coarse resolution, and then revert to a fine resolutionto refine the adjustment.

[0031] An advantage of the system 100 described herein (and of theinventive concepts embodied in the system 100) is that at the datadestination, the receiver does not need to synchronize with the mediathat transported the audio data. The receiver can therefore be designedessentially independently of the transport media, regardless of whetherthe media is T1/E1, optical fiber, RF, etc. The PLL and FIFO combinationadapts to the asynchronous nature of the transported data (in this case,audio data) after the transported data is removed from the transportmedia. Further, the inventive concepts described herein are not limitedto transporting audio data, and may also be used to transport many othertypes of digital data known in the art.

[0032] A three page supplemental document entitled “Rate Adaptation” isattached. This document describes, at a relatively high level, theunderlying concepts of the system 100 shown in FIG. 2.

[0033] The invention may be embodied in other specific forms withoutdeparting from the spirit or essential characteristics thereof. Thepresent embodiments are therefore to be considered in respects asillustrative and not restrictive, the scope of the invention beingindicated by the appended claims rather than by the foregoingdescription, and all changes which come within the meaning and range ofthe equivalency of the claims are therefore intended to be embracedtherein.

What is claimed is:
 1. A system for adapting a transmission rate of anaudio signal transmitted over a time division multiplexed communicationslink from a source to a destination, comprising: a first variable-depthstorage component for receiving and temporarily storing the audiosignal; a first processor for selectively withdrawing the audio signalfrom the first variable-depth storage component, for encapsulating theaudio signal into packets compatible with the time division multiplexedcommunication link, and for transmitting the packets over the timedivision multiplexed communications link; a second processor forreceiving the packets from the time division multiplexed communicationslink, for removing the audio signal from the encapsulating packets, andfor depositing the audio signal into a second variable-depth storagecomponent; wherein the first processor selectively withdraws the audiosignal as a predetermined function of an amount of data in the firstvariable-depth storage component, and the second processor selectivelywithdraws the audio signal as a predetermined function of an amount ofdata in the second variable-depth storage component.
 2. A systemaccording to claim 1, wherein the first variable-depth storage componentincludes a FIFO.
 3. A system according to claim 1, wherein the secondvariable-depth storage component includes a FIFO.
 4. A system accordingto claim 1, wherein the audio signal includes an AES/EBU digital audiosignal.
 5. A system according to claim 1, wherein the first processorinternally implements the first variable-depth storage component, andthe second processor internally implements the second variable-depthstorage component.
 6. A system according to claim 1, the secondprocessor being clocked by a clocking signal derived from a signalsource having a frequency which is variable according to a controlsignal from the second processor, wherein the second processor monitorsthe second variable-length storage component and adjusts the controlsignal so as to maintain the second variable-depth storage componentapproximately half full.
 7. A system according to claim 6, wherein thesignal source includes a PLL.
 8. A system according to claim 6, furtherincluding a signal conditioner for receiving a signal from the signalsource and producing a clocking signal therefrom.
 9. A system accordingto claim 8, wherein the signal conditioner is implemented with a FPGAdevice.
 10. A system according to claim 1, the first processor beingclocked by a clocking signal derived from a signal source having afrequency which is variable according to a control signal derived fromthe time division multiplexed communications link.
 11. A method ofadapting a transmission rate of an audio signal transmitted over a timedivision multiplexed communications link from a source to a destination,comprising: encapsulating the audio signal within the time divisionmultiplexed communications link; unpacking the audio signal from theencapsulation within the time division multiplexed communications link;and, varying the transmission rate of the audio signal at thedestination, after unpacking from the time division multiplexedcommunications link.
 12. A method of adapting a transmission rate of anaudio signal transmitted over a time division multiplexed communicationslink from a source to a destination, comprising: receiving andtemporarily storing the audio signal in a first variable-depth storagecomponent; selectively withdrawing the audio signal from the firstvariable-depth storage component as a predetermined function of anamount of data in the first variable-depth storage component,encapsulating the audio signal into packets compatible with the timedivision multiplexed communication link, and transmitting the packetsover the time division multiplexed communications link; receiving thepackets from the time division multiplexed communications link, removingthe audio signal from the encapsulating packets, and depositing theaudio signal into a second variable-depth storage component, andselectively withdrawing the audio signal as a predetermined function ofan amount of data in the second variable-depth storage component. 13 Amethod according to claim 12, further including implementing the firstvariable-depth storage component with a FIFO.
 14. A method according toclaim 12, further including implementing the second variable-depthstorage component with a FIFO.
 15. A method according to claim 12,further including adapting the audio signal includes as an AES/EBUdigital audio signal.
 16. A method according to claim 12, furtherincluding implementing the first variable-depth storage component andthe second variable-depth storage component in code, running on one ormore processors.
 17. A system for adapting a transmission rate of anaudio signal transmitted over a time division multiplexed communicationslink from a source to a destination, comprising: an encapsulator,including a first variable-depth storage component for receiving andtemporarily storing the audio signal, wherein the encapsulator (i)selectively withdraws the audio signal from the first variable-depthstorage component, (ii) encapsulates the audio signal into packetscompatible with the time division multiplexed communication link, and(iii) transmits the packets over the time division multiplexedcommunications link; and, an extractor for receiving the packets fromthe time division multiplexed communications link, for removing theaudio signal from the encapsulating packets, and for depositing theaudio signal into a second variable-depth storage component; wherein theencapsulator selectively withdraws the audio signal as a predeterminedfunction of an amount of data in the first variable-depth storagecomponent, and the extractor selectively withdraws the audio signal as apredetermined function of an amount of data in the second variable-depthstorage component.
 18. A system according to claim 17, wherein the firstvariable-depth storage component and the second variable-depth storagecomponent each include a FIFO.
 19. A system for adapting a transmissionrate of an AES/EBU digital audio signal transmitted over a time divisionmultiplexed communications link from a source to a destination,comprising: a first FIFO for receiving and temporarily storing theAES/EBU digital audio signal; a first processor for selectivelywithdrawing the AES/EBU digital audio signal from the first FIFO, forencapsulating the AES/EBU digital audio signal into packets compatiblewith the time division multiplexed communication link, and fortransmitting the packets over the time division multiplexedcommunications link; a second processor for receiving the packets fromthe time division multiplexed communications link, for removing theAES/EBU digital audio signal from the encapsulating packets, and fordepositing the AES/EBU digital audio signal into a second FIFO, thesecond processor being clocked by a clocking signal derived from asignal source having a frequency which is variable according to acontrol signal from the second processor, wherein the second processormonitors the second FIFO and adjusts the control signal so as tomaintain the second FIFO approximately half full. wherein the firstprocessor selectively withdraws the AES/EBU digital audio signal as apredetermined function of an amount of data in the first FIFO, and thesecond processor selectively withdraws the AES/EBU digital audio signalas a predetermined function of an amount of data in the second FIFO. 20.A method of adapting a transmission rate of an AES/EBU digital audiosignal transmitted over a time division multiplexed communications linkfrom a source to a destination, comprising: receiving and temporarilystoring the AES/EBU digital audio signal in a first FIFO; selectivelywithdrawing the AES/EBU digital audio signal from the first FIFO as apredetermined function of an amount of data in the first FIFO,encapsulating the AES/EBU digital audio signal into packets compatiblewith the time division multiplexed communication link, and transmittingthe packets over the time division multiplexed communications link;receiving the packets from the time division multiplexed communicationslink, removing the AES/EBU digital audio signal from the encapsulatingpackets, and depositing the AES/EBU digital audio signal into a secondFIFO, and selectively withdrawing the AES/EBU digital audio signal as apredetermined function of an amount of data in the second FIFO.